Communication through the Internet is based on the Internet Protocol (IP). The Internet is a packet-switched network versus the more traditional circuit switched voice network. The routing decision regarding an IP packet's next hop is made on a hop-by-hop basis. The full path followed by a packet is usually unknown to the transmitter, but it can be determined after the fact.
Packet loss over the Internet has been shown to be highly correlated. If packet N is dropped, there is a high probability that packet N+1 will also be dropped. Kostas, Borella, et al. “Real-Time voice over Packet-Switched Networks,” IEEE Network, Jan./Feb. 1998.
Voice over Packet/Voice over IP (VOP/VOIP) solutions contend with bandwidth, delay jitter, delay, and packet loss issues. Voice codecs provide some form of packet loss-concealment, or packet loss mitigation, by utilizing coding and packetization schemes over a single flow. These schemes employ some combination of buffering and redundancy (forward error correction (FEC), multi-rate encoding, etc.). The redundant portion of the coding is inserted into some other packet in the single flow. Such coding provides an intra/inter packet diversity gain that can help conceal packet loss, delay, and delay jitter.
TCP is a transport layer 4 protocol and IP is a network layer 3 protocol. IP is unreliable in the sense that it does not guarantee that a sent packet will reach its destination. TCP is provided on top of IP to guarantee packet delivery by tagging each packet. Lost or out of order packets are detected and then the source supplies a responsive retransmission of the packet to destination. Because the packet retransmission process takes significant time, TCP may not satisfactorily solve problems in quality transmission of audio, also known as Voice over IP or VOP (Voice over Packet), and video and other media where maximum tolerable packet delay is not high.
UDP is a transport layer 4 protocol that eliminates the overhead of the retransmission mechanism of TCP but does not make a guarantee that every sent packet will be received. The interface to command the use of either UDP or TCP is very similar.
DIFFSERV is a class of service initiative spearheaded by the IETF Internet Engineering Task Force. Class A service will be better than Class B and Class B is better than Class C. IETF has tried to improve QoS (quality of service). QoS categories are mapped by a policy mechanism to the DIFFSERV class of service categories. However, the policies of different subnetworks (domains) of different companies do vary. As a packet traverses the Internet it typically crosses various companies' domains. These companies require and need to track payment for various classes of services, so permission and authentication mechanisms are keyed to the various service policies of the companies.
RSVP is an edge-network protocol which requires that every intermediate router understand RSVP. Over long distances typical of the Internet this condition does not usually pertain. RSVP is utilized in enterprise networks.